Just a follow up with the confirmed resolution. I had copied the FS Directory entries for these Polycom Phones from a CISCO phone setup. That setup specified: <param name="NDLB-force-rport" value="true"/> Ken adviced to change that for Polycom Phones to: <param name="NDLB-force-rport" value="safe"/> The customer emailed me today to say the problem has been resolved. Thanks for all who sent their thoughts. Extra thanks to Ken. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, December 05, 2013 9:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP Trace Question / NAT one way audio problem. Re-registering every 23 seconds isnt that big of a deal if the phones are behind nat, sometimes this is needed due to the way NAT happens on UDP with various routers. There is no way to track the start/end of the session w/out using an ALG. I have seen some NAT routers require a re-reg interval around 15 seconds which is a bit excessing On the issue w/ the non-working ones is you'll notice the rport on the one that's working... On the ones that arent working they arent specifying rport... If those are the polycoms look for NDLB force rport setting and set it to safe... That's a special mode for the polycoms... Check the wiki for more NAT handling examples K On 12/5/13 8:36 PM, "Sean P Devoy" <sdevoy@bizfocused.com> wrote: [cid:image001.gif@01CEFD76.90387370] Hi all, I seem to have fallen off the list! No mail since 7/26/2013. Have I made someone mad?? :) I have also been blissfully running with ISSUES what so ever since that time. I have a NATed client with PERIODIC one way audio. The client end has a Cisco 220W router with SIP ALG disabled. The phones are CISCO 504Gs and Polycom 330s(?), basciallt the same as everywhere else we have phones. I did a sofia global sip trace on for a while and captured the output. After writing an app to sort those sip packets into logical stream files, I dug in to the SIP conversations. This has left me with som questions for the people who know these things: 1. I found all my phones are REGISTERing every 23 seconds! That seems absurd TO ME, so where have I screwed that up/where do I set it and what is a reasonable number? 2. The only difference I see in the SIP packets from my working phone and these non-working ones is of course NAT related. The working phone's REGISTER looks like this: recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203: ------------------------------------------------------------------------ REGISTER sip: <MYDOMAIN.COM> SIP/2.0 Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport But the failing phones look like this: recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691: ------------------------------------------------------------------------ REGISTER sip: <MYDOMAIN.COM> SIP/2.0 Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431 The response is still sent to [AA.BB.CC.38]:5104 I guess first, IS THAT A PROBLEM? So, can anyone share any insight into the CISCO 220W config that I might be missing? Why would it only be a periodic problem? Could this be a problem where one end is using a wider range of rports than the other supports? Thanks, Sean ________________________________ <http://www.avast.com/> This email is free from viruses and malware because avast! Antivirus <http://www.avast.com/> protection is active. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch
